November 2013 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Copy all sip headers between two legs
by Mickael Hubert 05 Dec '13

05 Dec '13

hi list, I want copy all sip headers between two legs, but I don't obtain a good result. example: P-Asserted-identity, P-Preferred-Identity, Diversion, etc ... aren't copied for leg B. I have found "sip_copy_custom_headers=true" but it's just for X-Header. Actually, I use this code below, but it is not easy :( *<condition field="${sip_P-Asserted-Identity}" expression="\+(\d+)" break="never"> <action application="log" data="LOG Modifying PAI sip:+33xxxxxxxxx@DOMAIN" /> <action application="export" data="sip_h_P-Asserted-Identity=<sip:+$1@3008LI.hexavoip.fr <1(a)3008LI.hexavoip.fr>>" />* Have you an idea for me ? Thanks in advance -- Cordialement HUBERT Mickaël Ingénieur VOIP - Hexanet --

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Getting FreeSWITCH to use RTP port + 1 for RTCP
by James Mortensen 04 Dec '13

04 Dec '13

Hello, I've searched the list and documentation, and it's not yet clear if FreeSWITCH is able to send candidates in the same manner as we were doing with Asterisk 1.5 where it sends the RTCP to the RTP port +1. In other words, I'd see candidates that looked like this: a=candidate:3441596188 1 udp 659136 10.3.1.1 20622 typ host generation 0 a=candidate:3441596188 1 udp 659136 50.18.X.X 20622 typ srflx generation 0 a=candidate:3441596188 2 udp 659136 10.3.1.1 20623 typ host generation 0 a=candidate:3441596188 2 udp 659136 50.18.X.X 20623 typ srflx generation 0 For instance, if the RTP port was 20622, the RTCP port would be 20623. We're getting no audio when trying to make Chrome to Android and Android to Android WebRTC calls, and we thought we'd try a different RTCP method as we did have it working with Asterisk but want to use FreeSWITCH instead. I've checked out the latest FreeSWITCH code from the master branch. Not sure if FreeSWITCH can do this yet, but if anyone knows that would be helpful. Thank you! James

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Questions for an operator interconnect
by James Lotery 04 Dec '13

04 Dec '13

Hello, we are planning to interconnect with a national operator using FS of course ;) Amongst other things they are the following for which I haven't found answers to. If anyone knows I'd be grateful for an answer: 1. Can we turn off INFO message 2. Are Tel URIs supported (From what I can tell the answer is no, but need confirmation) Thanks in advance. James

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Re: [Freeswitch-users] mod_opus codec on Windows
by Jeff Leung 04 Dec '13

04 Dec '13

Perfect, I'll try it out when I have the time to do so. Thanks for going to places where I wouldn't have normally went through ;) Peter Olsson <peter(a)olssononline.se> wrote: I just wanted to notify that there is a patch available for this now (FS-5704). It has not yet been commited to git though. If anyone want to test it out, please apply the patch manually. /Peter 9 jul 2013 kl. 08:09 skrev Peter Olsson <peter(a)olssononline.se>: I think so too. I've started on a complete patch for the Windows build, hopefully I will have some time during this week to complete it. I'll submit this to Jira when finished. 2013/7/8 Michael Jerris <mike(a)jerris.com> > Are you sure symbol visibility is a problem even if these are in different > dll's ? I think this shouldn't be an issue. > > Mike > > On Jul 6, 2013, at 12:10 PM, Jeff Leung <jleung(a)v10networks.ca> wrote: > > The project was within FreeSWITCH, but the problem is because the way how > the opus lib builds, it includes silk and celt and I wasn’t going to have > FS to build all of that in tree because I was afraid the symbols from OPUS > would conflict with the symbols in CELT and SILK.**** > > *From:* freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch- > users-bounces(a)lists.freeswitch.org] *On Behalf Of *Peter Olsson > *Sent:* Saturday, July 6, 2013 2:02 AM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] mod_opus codec on Windows**** > ** ** > Ah, ok. So you did the build of the lib outside fs, and then just pointed > to the built library?**** > ** ** > I'll start working to get this integrated in the Windows build, by > downloading the lib source files, unpacking them, and then building them.* > *** > ** ** > I will submit a Jira as soon as I have some progress. I might get back to > you with further questions.**** > ** ** > /Peter**** > > ** ** > 2013/7/6 Jeff Leung <jleung(a)v10networks.ca>**** > The project file is nothing more than having mod_opus to point to an out of > tree directory where the actual opus binary libs are built. > > The binaries I did was basically a bunch of hacks, there is quite some work > needed to be done with integrating the build process of opus with > FreeSWITCH's VS solution file > > > -----Original Message----- > > From: freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch- > > users-bounces(a)lists.freeswitch.org] On Behalf Of Peter Olsson > > Sent: Saturday, July 6, 2013 12:07 AM > > To: FreeSWITCH Users Help > > Subject: Re: [Freeswitch-users] mod_opus codec on Windows > > > > Good work! Could you also please file a Jira, and upload the project > files > and > > modifications that you've made for this to work? I'd be happy to assist > and > > get this commited. > > > > /Peter > > > > 6 jul 2013 kl. 08:20 skrev Jeff Leung <jleung(a)v10networks.ca>: > > > > > As I heard on last Wednesday's conference call, mod_opus isn't > "available" > > > on Windows, basically meaning that FreeSWITCH users can't fully > > > implement WebRTC on Windows Platforms just yet. > > > > > > I have taken the steps to have the codec successfully build on this > > > platform; However it's a pain in the neck to build as the opus library > > > generates multiple libs which includes CELT and SILK codecs (Go > figure). > > > > > > Shoot me a message off-list if you would like to use the binary I've > > > built for Win64 platforms. > > > > > > Before anybody asks for a 32-bit binary, the short answer is going to > be > no. > > > I won't be providing a 32-bit binary as I intend this binary to be > > > used in a production environment. > > > > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting(a)freeswitch.org > http://www.freeswitchsolutions.com > > FreeSWITCH-powered IP PBX: The CudaTel Communication Server > http://www.cudatel.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > >

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Max registrations per extension
by Rafael Santana 03 Dec '13

03 Dec '13

Hi, I am trying to limit the number of registrations using a specific extension (e.g 1000) to only 1 per time. To do it I am setting the "max-registrations-per-extension" variable to "1". The problem is that when I do it, the extension 1000 becomes forbidden to make calls :( Can someone explain it to me? Is it a bug? -- Rafael Santana Oliveira Mestre em Ciência da Computação

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Detecting and blocking fax calls
by Everyday User 02 Dec '13

02 Dec '13

We receive an inordinate amount of spam fax calls to FreeSwitch that we do not need to handle for any reason. I came across this method, https://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_fax_detect, to hopefully detect and drop these calls. Is there any danger of this method mistaking any live calls and dropping those on accident or is there a better method to accomplish this? Thanks in advance.

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How to use mod_lcr with lrn and destination number?
by Victor Chukalovskiy 02 Dec '13

02 Dec '13

Hi, What is the syntax to use mod_lcr and route based on lrn? How to pass LRN to mod_lcr together with the destination number? Is it something like *lcr <destination number> <profile> <lrn=lrn_number>*? This section of WiKi is very scarce and does not answer the question: http://wiki.freeswitch.org/wiki/Mod_lcr#LRN I'll improve the WiKi once I know the answer myself :-) Thank you, Victor

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503 service unavalible
by Miha 02 Dec '13

02 Dec '13

HI, I need a little help figuring out why FS sends 503. I happens when 302 happens and opensips relay another call throught FS. It started to happening why I used in a brdige dns srv domain. All other trasfers and call a lookig good and wokirng. p.s.: 302 is handled by opensips. tnx here is a pastebin: http://pastebin.freeswitch.org/21677 And here is just sip trace: 1028.883730 UAC_PUBLIC_IP -> OPENSIPS_IP SIP/SDP Request: INVITE sip:014310000@dns_srv.domain.com, with session description 1028.888153 OPENSIPS_IP -> UAC_PUBLIC_IP SIP Status: 100 Giving a try 1028.888248 OPENSIPS_IP -> FS_IP SIP/SDP Request: INVITE sip:38614310000@dns_srv.domain.com, with session description 1028.889756 FS_IP -> OPENSIPS_IP SIP Status: 100 Trying 1028.916452 84.41.126.2 -> OPENSIPS_IP SIP Request: NOTIFY sip:dns_srv.domain.com 1028.917245 OPENSIPS_IP -> 84.41.126.2 SIP Status: 407 Proxy Authentication Required 1028.926470 FS_IP -> OPENSIPS_IP SIP/SDP Request: INVITE sip:38614310000@dns_srv.domain.com, with session description 1028.934352 OPENSIPS_IP -> FS_IP SIP Status: 100 Giving a try 1028.934438 OPENSIPS_IP -> UAC_PUBLIC_IP SIP/SDP Request: INVITE sip:38614310000@UAC_PUBLIC_IP:8604, with session description 1028.954083 UAC_PUBLIC_IP -> OPENSIPS_IP SIP Status: 100 Trying 1028.956366 84.41.126.2 -> OPENSIPS_IP SIP Request: NOTIFY sip:dns_srv.domain.com 1028.956998 OPENSIPS_IP -> 84.41.126.2 SIP Status: 484 Address Incomplete 1028.957910 UAC_PUBLIC_IP -> OPENSIPS_IP SIP Status: 302 Moved Temporarily 1028.958296 OPENSIPS_IP -> UAC_PUBLIC_IP SIP Request: ACK sip:38614310000@UAC_PUBLIC_IP:8604 1028.976068 OPENSIPS_IP -> FS_IP SIP/SDP Request: INVITE sip:30238651357952@dns_srv.domain.com, with session description 1028.977116 FS_IP -> OPENSIPS_IP SIP Status: 100 Trying 1029.036370 FS_IP -> OPENSIPS_IP SIP Status: 503 Service Unavailable 1029.036849 OPENSIPS_IP -> FS_IP SIP Request: ACK sip:30238651357952@dns_srv.domain.com 1029.037055 OPENSIPS_IP -> FS_IP SIP Status: 500 Service Unavailable

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More Nibblebill questions
by Andrew Cassidy 02 Dec '13

02 Dec '13

So this is starting to get complicated and it looks like I might have to write some lua to do this... So, say I want to bridge a freephone number to both a landline and a mobile in parallel. All 3 potential legs need different call rates, only 2 will be used when the call is picked up, of course, and I ideally need the CDR and cost for each leg separate. Here's an idea of what I have so far: <action application="set" data="nibble_rate=0.01" /> <action application="bridge" data="[nibble_rate=0.01]sofia/gateway/x/landline,[nibble_rate=0.06]sofia/gateway/x/mobile" /> This doesn't work, and neither does setting the variables nolocal. Can this be done in this way? I can probably work around this by exporting the incoming rate as nolocal and the outgoing rate on the incoming channel. Confusing but might actually work. -- *Andrew Cassidy BSc (Hons) MBCS SSCA* Managing Director *T <info(a)cassidywebservices.co.uk> *03300 100 960 *F <info(a)cassidywebservices.co.uk> *03300 100 961 *E <info(a)cassidywebservices.co.uk> *andrew(a)cassidywebservices.co.uk *W <info(a)cassidywebservices.co.uk> *www.cassidywebservices.co.uk

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reinstall module
by Joli Martinez 02 Dec '13

02 Dec '13

Hello I have the dignaling module is damaged. I need to reinstall it. How can I recompiile it again? thanks,

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